The present disclosure relates to digital audio files, and to systems and methods for comparing the contents of two or more such files.
Digital-based electronic media formats have become widely accepted. The development of faster computer processors, high-density storage media, and efficient compression and encoding algorithms have led to an even more widespread implementation of digital audio media formats in recent years. Digital compact discs (CDs) and digital audio file formats, such as MP3 (MPEG Audio—layer 3) and WAV, are now commonplace. Some of these formats store the digitized audio information in an uncompressed fashion while others feature compression. The ease with which digital audio files can be generated, duplicated, and disseminated also has helped increase their popularity.
Audio information can be detected as an analog signal and represented using an almost infinite number of electrical signal values. An analog audio signal is subject to electrical signal impairments, however, that can negatively affect the quality of the recorded information. Any change to an analog audio signal value can result in a noticeable defect, such as distortion or noise. Because an analog audio signal can be represented using an almost infinite number of electrical signal values, it is also difficult to detect and correct defects. Moreover, the methods of duplicating analog audio signals cannot approach the speed with which digital audio files can be reproduced. These and many other problems associated with analog audio signals can be overcome, without a significant loss of information, simply by digitizing the audio signals.
FIG. 1 presents a portion of an analog audio signal 10. The amplitude of the analog audio signal 10 is shown with respect to the vertical axis 12 and the horizontal axis 14 indicates time. In order to digitize the analog audio signal 10, the waveform 16 is sampled at periodic intervals, such as at a first sample point 18 and a second sample point 20. A sample value representing the amplitude of the waveform 16 is recorded for each sample point. If the sampling rate is less than twice the frequency of the waveform being sampled, the resulting digital signal will be substantially identical to the result obtained by sampling a waveform of a lower frequency. As such, in order to be adequately represented, the waveform 16 must be sampled at a rate greater than twice the highest frequency that is to be included in the reconstructed signal. To ensure that the waveform is free of frequencies higher than one-half of the sampling rate, which is also known as the Nyquist frequency, the audio signal 10 can be filtered prior to sampling. Therefore, in order to preserve as much audible information as possible, the sampling rate should be sufficient to produce a reconstructed waveform that cannot be differentiated from the waveform 16 by the human ear.
The human ear generally cannot detect frequencies greater than 16-20 kHz, so the sampling rate used to create an accurate representation of an acoustic signal should be at least 32 kHz. For example, compact disc quality audio signals are generated using a sampling rate of 44.1 kHz. Once the sample value associated with a sample point has been determined, it can be represented using a fixed number of binary digits, or bits. Encoding the infinite possible values of an analog audio signal using a finite number of binary digits will almost necessarily result in the loss of some information. Because high-quality audio is encoded using up to 24-bits per sample, however, the digitized values closely approximate the original analog values. The digitized values of the samples comprising the audio signal can then be stored using a digital-audio file format.
The technique by which analog audio information is digitized is flexible and can be implemented in many different ways. For example, an analog signal can be sampled at many different locations and the sample values can be quantized to varying degrees of accuracy. Because an analog audio signal is represented digitally using only discrete samples of the constant waveform and because the continuously varying signal level is quantized into finite values, two digital audio files representing the same analog audio signal can be comprised of very different bits. Also, the bits representing an audio signal can be stored using different file formats, such as .DV or .MOV. Because such file formats can store portions of an audio signal in different locations within a file, it can be impossible to recognize the commonality between two identical audio signals.
FIG. 2 presents an analog audio signal 50 that is digitized by sampling the waveform 52 at a plurality of points. For example, the waveform 52 can be sampled at the points associated with solid lines, including points 54 and 56. Alternatively, the waveform 52 can be sampled at the points associated with dashed lines, including points 58 and 60. Although the sampling frequency associated with the solid lines and the dashed lines is the same, samples are taken at different points in time along the waveform 52. If the sampling frequency associated with the solid lines and the dashed lines is equal to or greater than the Nyquist rate, the waveform 52 can be accurately reconstructed from either of the resulting digital representations. Therefore, the waveform reconstructed using the sample points associated with the solid lines, including the points 54 and 56, will be substantially identical to the waveform reconstructed using the sample points associated with the dashed lines, including the points 58 and 60. Still, the bits associated with the respective sample points can be very different because those sample points occur at different points in time.
A similar result occurs if separate digital audio files are created by sampling the waveform 52 at different rates. For example, a first digital audio file can be generated by sampling the waveform 52 at a sampling rate of 44 kHz and a second digital audio file can be generated by sampling the waveform 52 at a sampling rate of 45 kHz. If all other factors are identical, the reconstructed waveform produced from the first digital audio file will be substantially identical to the reconstructed waveform produced from the second digital audio file. The bits of the first digital audio file, however, will differ from the bits of the second digital audio file because the waveform 52 is sampled at different points.
Additionally, different digital representations of the waveform 52 can result from a single set of samples if the sample values are quantized using a different number of bits. For example, if the sample values are quantized using 20-bits to generate a first digital audio file and 24-bits to generate a second digital audio file, the first and second files will differ significantly at the bit level. Similarly, differing digital representations of an identical waveform also can be generated by applying differing compression techniques.
As discussed above, an analog audio signal can be digitized in accordance with a variety of techniques and methods. Therefore, it is possible for a large number of distinct binary representations to produce identical, or substantially identical, audio signals. In order to determine whether the audio signals associated with two digital audio files are identical, it is thus necessary to compare the files using some measure other than the bits that comprise those files. For example, a developer of audio signal processing hardware or software can find it necessary to compare two or more digital audio files, such as a first file that represents an audio signal after it has been processed and a second file that represents a control sample. The control sample can be any file that represents a known audio signal, such as a file representing the audio signal prior to processing or a reference signal that is an accurate representation of the desired audio signal after processing. The comparison can thus be used to identify any discontinuities that might have been introduced by the processing operation.